Introduction to Digital Communication:

Data communication or digital communications, including data transmission and

data reception, is the transfer and reception of data in the form of a digital
bitstream or a digitized analog signal transmitted over a point-to-point or point-
to-multipoint communication channel.
Digital refers to the discrete time-varying signal.

Block diagram of Digital Communication System:

The block diagram description:

SOURCE ENCODER / DECODER: The Source encoder ( or Source coder)
converts the input i.e. symbol sequence into a binary sequence of 0’s and 1’s by
assigning code words to the symbols in the input sequence. For eg. :-If a source
set is having hundred symbols, then the number of bits used to represent each
symbol will be 7 because 27=128 unique combinations are available. The
important parameters of a source encoder are block size, code word lengths,
average data rate and the efficiency of the coder (i.e. actual output data rate
compared to the minimum achievable rate) At the receiver, the source decoder
converts the binary output of the channel decoder into a symbol sequence. The
decoder for a system using fixed – length code words is quite simple, but the
decoder for a system using variable – length code words will be very complex.
Aim of the source coding is to remove the redundancy in the transmitting
information, so that bandwidth required for transmission is minimized. Based on
the probability of the symbol code word is assigned. Higher the probability,
shorter is the code word. Ex: Huffman coding.
CHANNEL ENCODER / DECODER: Error control is accomplished by the
channel coding operation that consists of systematically adding extra bits to the
output of the source coder. These extra bits do not convey any information but
helps the receiver to detect and / or correct some of the errors in the information
bearing bits. There are two methods of channel coding:
1. Block Coding: The encoder takes a block of ‘k’ information bits from the
source encoder and adds ‘r’ error control bits, where ‘r’ is dependent on ‘k’ and
error control capabilities desired.
2. Convolution Coding: The information bearing message stream is encoded in
a continuous fashion by continuously interleaving information bits and error
control bits.
The Channel decoder recovers the information bearing bits from the coded binary
stream. Error detection and possible correction is also performed by the channel
decoder. The important parameters of coder / decoder are: Method of coding,
efficiency, error control capabilities and complexity of the circuit.
MODULATOR: The Modulator converts the input bit stream into an electrical
waveform suitable for transmission over the communication channel. Modulator
can be effectively used to minimize the effects of channel noise, to match the
frequency spectrum of transmitted signal with channel characteristics, to provide
the capability to multiplex many signals.
DEMODULATOR: The extraction of the message from the information bearing
waveform produced by the modulation is accomplished by the demodulator. The
output of the demodulator is bit stream. The important parameter is the method
of demodulation.
CHANNEL: The Channel provides the electrical connection between the source
and destination. The different channels are: Pair of wires, Coaxial cable, Optical
fibre, Radio channel, Satellite channel or combination of any of these.
The communication channels have only finite Bandwidth, non-ideal frequency
response, the signal often suffers amplitude and phase distortion as it travels over
the channel. Also, the signal power decreases due to the attenuation of the
channel. The signal is corrupted by unwanted, unpredictable electrical signals
referred to as noise. The important parameters of the channel are Signal to Noise
power Ratio (SNR), usable bandwidth, amplitude and phase response and the
statistical properties of noise.
Advantages of digital communication
• Digital circuits that are relatively affordable can be employed.
• Data encryption is used to protect privacy.
• Data from Voice, Video, and Data sources may all be combined and sent via a
single digital transmission system.
• Unlike analog systems, noise does not build from repeater to repeater in long-
distance networks.
• Errors detected are minor, even when the received signal contains a lot of noise,
i.e. the S/N Ratio in Digital Systems is high.
• Errors are frequently rectified by using error correction coding.

Disadvantages of digital communication

• More bandwidth is necessary for digital systems than for analog systems and
synchronization is required.

Merits of Digital Communication

1. The effect of distortion, noise and interference is less in a digital

communication system. This is because the disturbance must be large enough to
change the pulse from one state to the other.

2. Regenerative repeaters can be used at fixed distance along the link, to identify
and regenerate a pulse before it is degraded to an ambiguous state.

3. Digital circuits are more reliable and cheaper compared to analog circuits.

4. The Hardware implementation is more flexible than analog hardware because

of the use of microprocessors, VLSI chips etc.
5. Signal processing functions like encryption, compression can be employed to
maintain the secrecy of the information.

6. Error detecting and Error correcting codes improve the system performance by
reducing the probability of error.

7. Combining digital signals using TDM is simpler than combining analog signals
using FDM. The different types of signals such as data, telephone, TV can be
treated as identical signals in transmission and switching in a digital
communication system.

8. We can avoid signal jamming using spread spectrum technique.

Demerits of Digital Communication:

1. Large System Bandwidth:- Digital transmission requires a large system

bandwidth to communicate the same information in a digital format as compared
to analog format.

2. System Synchronization:- Digital detection requires system synchronization

whereas the analog signals generally have no such requirement.

Sampling theorem

The sampling theorem is an important aid in the design and analysis of

communication systems involving the use of continuous time functions of finite
bandwidth. Also, the data transmission in the form of digital signal offers various
advantages, such as high efficiency, fast speed, low cost, low interference, low
distortion, and high security. Hence, sampling is essential to improve the quality
and transmission ability of the signals over the communication channel.

Statement: A continuous time signal can be represented in its samples and can
be recovered back when sampling frequency fs is greater than or equal to the twice
the highest frequency component of message signal. i. e. fs≥2fm.
Proof: Consider a continuous time signal x(t). The spectrum of x(t) is a band
limited to fm Hz i.e. the spectrum of x(t) is zero for |ω|>ωm.

Sampling of input signal x(t) can be obtained by multiplying x(t) with an impulse
train δ(t) of period Ts. The output of multiplier is a discrete signal called sampled
signal which is represented with y(t) in the following diagrams:

Here, the sampled signal takes the period of impulse. The process of sampling
can be explained by the following mathematical expression:

Sampled signal y(t)=x(t).δ(t)......(1)

To reconstruct x(t), the input signal spectrum X(ω) must be recovered from
sampled signal spectrum Y(ω), which is possible when there is no overlapping
between the cycles of Y(ω).

Possibility of sampled frequency spectrum with different conditions is given by

the following diagrams:

Aliasing Effect

The overlapped region in case of under sampling represents aliasing effect, which
can be removed by

• considering fs >2fm
• By using anti aliasing filters.

It causes different signals to become indistinguishable when sampled. It also

often refers to the distortion or artifacts that result when a signal reconstructed
from samples is different from the original continuous signal.
 • Aliasing can have a number of negative effects on the performance of a
communication system, including:
1. Reduced signal-to-noise ratio: Aliasing can introduce additional noise
into the reconstructed signal, which can degrade the signal-to-noise ratio
and make it more difficult to accurately recover the original signal.
2. Interference with other signals: Aliased signals can overlap with other
signals in the frequency domain, causing interference and reducing the
quality of the transmitted or received signal.
3. Reduced bandwidth efficiency: Aliasing can cause the bandwidth of a
signal to appear wider than it actually is, which can reduce the overall
efficiency of the communication system.
• To avoid these problems, it is important to use a high enough sampling
rate when digitizing a signal for transmission or storage in a
communication system. This is known as the Nyquist rate, which states
that the sampling rate must be at least twice the highest frequency present
in the original signal.
There are several ways to reduce or eliminate aliasing when sampling a signal:

1. Use a higher sampling rate: Increasing the sampling rate above the
Nyquist rate will help to reduce aliasing. This is because the higher rate
allows for more samples to be taken of the original signal, which can
more accurately capture its details.
2. Use an anti-aliasing filter: An anti-aliasing filter is a low-pass filter that
removes high-frequency components from the signal before it is sampled.
This helps to prevent the aliasing of high-frequency components, which
would otherwise be incorrectly interpreted as lower frequencies during
reconstruction.
3. Use oversampling: Oversampling involves taking more samples of the
signal than is strictly necessary according to the Nyquist rate. The extra
samples can then be averaged or filtered to reduce aliasing.

There are three types of sampling techniques:

• Impulse sampling.
• Natural sampling.
• Flat Top sampling.

Impulse Sampling
Also called ideal sampling, impulse sampling can be performed by multiplying
input signal x(t) with impulse train Σn=−∞δ(t−nT) of period 'T'. Here, the
amplitude of impulse changes with respect to amplitude of input signal x(t). The
output of sampler is given by Σn=−∞δ(t−nT)

Natural Sampling

Natural sampling is similar to impulse sampling, except the impulse train is

replaced by pulse train of period T. i.e. input signal x(t) is multiplied with the
pulse train Σn=−∞ P(t−nT) as shown.

Flat Top Sampling

During transmission, noise is introduced at top of the transmission pulse which

can be easily removed if the pulse is in the form of flat top. Here, the top of the
samples are flat i.e. they have constant amplitude. Hence, it is called as flat top
sampling or practical sampling. Flat top sampling makes use of sample and hold
circuit.
Theoretically, the sampled signal can be obtained by convolution of rectangular
pulse p(t) with ideally sampled signal say yδ(t) as shown in the diagram:

i.e. y(t)=p(t)×yδ(t)......(1)

To get the sampled spectrum, consider Fourier transform on both sides for
equation 1

Y[ω]=F.T[P(t)×yδ(t)]

By the knowledge of convolution property,

Y[ω]=P(ω)Yδ(ω)
Here P(ω)= 2sinωT/ω

Nyquist Rate

It is the minimum sampling rate at which signal can be converted into samples
and can be recovered back without distortion.

Nyquist rate fN = 2fm hz

Quadrature Sampling of Band pass signals

{ Quadrature-sampling is the process of digitizing a continuous (analog)

bandpass signal and translating its spectrum to be centered at zero Hz

Bandpass sampling performs digitization and frequency translation in a single

process. In BPS, we can sample the signal at twice the information bandwidth
and remain consistent with the Nyquist sampling theorem.
Advantage of bandpass sampling-
Since the sampling time is more, this implies that the speed requirement is less.
Because of less speed requirement, it will store less number of samples compared
to low pass sampling. It will, therefore, decrease the memory
requirement because it stores fewer samples when compared to low-pass
sampling.}
This scheme represents a natural extension of the sampling of low – pass signals.
In this scheme, the band pass signal is split into two components, one is in-phase
component and other is quadrature component. These two components will be
low–pass signals and are sampled separately. This form of sampling is called
quadrature sampling. Let g(t) be a band pass signal, of bandwidth ‘2W’ centered
around the frequency, fc, (fc>W). The in-phase component, gI(t) is obtained by
multiplying g(t) with cos(2πfct) and then filtering out the high frequency
components. Parallelly a quadrature phase component is obtained by multiplying
g(t) with sin(2πfct) and then filtering out the high frequency components..The
band pass signal g(t) can be expressed as,

g(t) = gI(t). cos(2πfct) – gQ(t) sin(2πfct)

The in-phase, gI(t) and quadrature phase gQ(t) signals are low–pass signals,
having band limited to (-W < f < W). Accordingly each component may be
sampled at the rate of 2W samples per second.
Pulse-Amplitude Modulation (PAM):

Pulse-amplitude modulation is the simplest form of pulse modulation. It is

generated in much the same manner as analog-amplitude modulation. The timing
pulses are applied to a pulse amplifier in which the gain is controlled by the
modulating waveform. Since these variations in amplitude actually represent the
signal, this type of modulation is basically a form of AM. The only difference is
that the signal is now in the form of pulses. This means that PAM has the same
builtin weaknesses as any other AM signal -high susceptibility to noise and
interference. The reason for susceptibility to noise is that any interference in the
transmission path will either add to or subtract from any voltage already in the
circuit (signal voltage). Thus, the amplitude of the signal will be changed. Since
the amplitude of the voltage represents the signal, any unwanted change to the
signal is considered a SIGNAL DISTORTION. For this reason, pam is not often
used. When pam is used, the pulse train is used to frequency modulate a carrier
for transmission. Techniques of pulse modulation other than pam have been
developed to overcome problems of noise interference.
The advantages of pulse amplitude modulation include the following.

• It is a simple process for both modulation and demodulation.

• Transmitter and receiver circuits are simple and easy to construct.
• PAM can generate other pulse modulation signals and can carry the message
at the same time.
• The data can be transmitted quickly, efficiently, and effectively through usual
copper wires in high volume.
• The FM available is infinite; therefore the development of PAM can be done
frequently to permit enhanced data throughput over accessible networks.
• It is the simplest type of modulation
• For all types of digital modulation methods, it is the base and simple method
for both modulation & demodulation.
• For both the transmission as well as reception, it doesn’t require complex
circuitry. The circuit design of the Transmitter & receiver is very simple.
• This modulation can generate other types of pulse modulation signals & also
carries the message at the same time.

Disadvantages
The disadvantages of pulse amplitude modulation include the following.

• Bandwidth should be large for transmission PAM modulation.

• Noise will be great.
• Pulse amplitude signal varies so the power required for transmission will be
more.
• For transmitting PAM signal, BW must be large
• The frequency changes based on the message or modulating signal because of
these changes within the frequency of the signal, intrusions will be there.
• For this modulation, noise immunity is low as compared to other types. So it
is nearly equivalent to AM.
• Once pulse amplitude signal changes then the required power for transmission
is high and even to get the PAM, more power is necessary.

Applications of PAM
• It is used in Ethernet communication.
• It is used in many micro-controllers for generating control signals.
• It is used in Photo-biology.
• It is used as an electronic driver for LED lighting.
• PAM is used in the Ethernet network which is used to connect two systems &
used to transfer data among these systems. So PAM is used in Ethernet
communications.
• The control signals can be generated in various microcontrollers by using PAM
• This modulation technique is mostly used in digital data transmission &
applications changed by PCM &PPM. Mostly all phone modems which are
faster above 300 bit/s utilize QAM (quadrature amplitude modulation).
Time division multiplexing (TDM):

It is often practical to combine a set of low-bit-rate streams, each with a fixed and
pre-defined bit rate, into a single high-speed bit stream that can be transmitted
over a single channel. This technique is called time division multiplexing (TDM)
and has many applications, including wire line telephone systems and some
cellular telephone systems. The main reason to use TDM is to take advantage of
existing transmission lines. It would be very expensive if each low-bit-rate stream
were assigned a costly physical channel (say, an entire fiber optic line) that
extended over a long distance. The high-bit-rate channel can be divided into a
series of time slots, and the time slots can be alternately used by the three sources.
The three sources are thus capable of transmitting all of their data across the
single, shared channel. At the other end of the channel, the process must be
reversed (i.e., the system must divide the 192 Kbit/sec multiplexed data stream
back into the original three 64 Kbit/sec data streams, which are then provided to
three different users). This reverse process is called demultiplexing.

Choosing the proper size for the time slots involves a trade-off between efficiency
and delay. If the time slots are too small (say, one bit long) then the multiplexer
must be fast enough and powerful enough to be constantly switching between
sources (and the demultiplexer must be fast enough and powerful enough to be
constantly switching between users). If the time slots are larger than one bit, data
from each source must be stored (buffered) while other sources are using the
channel. This storage will produce delay. If the time slots are too large, then a
significant delay will be introduced between each source and its user. Some
applications, such as teleconferencing and videoconferencing, cannot tolerate
long delays.

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